Hearing instruments or hearing aids typically comprise a microphone amplification assembly which includes one or several microphones for receipt of incoming sound such as speech and music. The incoming sound is converted to an electric microphone signal or signals that are amplified and processed in a control and processing circuit of the hearing instrument in accordance with one or more preset listening program(s). These listening programs have typically been computed from a user's specific hearing deficit or loss for example expressed in an audiogram. An output amplifier of the hearing instrument delivers the processed microphone signal to the user's ear canal via a miniature speaker or receiver that may be housed in a casing of the hearing instrument together with the microphone or separately in an ear plug.
A hearing impaired person typically suffers from a loss of hearing sensitivity which loss is dependent upon both frequency and the level of the sound in question. Thus a hearing impaired person may be able to hear certain frequencies (e.g., low frequencies) as well as a normal hearing person, but unable to hear sounds with the same sensitivity as the non-hearing impaired person at other frequencies (e.g., high frequencies). Similarly, the hearing impaired person may be perceive loud sounds, e.g. above 90 dB SPL, with the same intensity as the non-hearing impaired person, but unable to hear soft sounds with the same sensitivity as the non-hearing impaired person. Thus, in the latter situation, the hearing impaired person suffers from a loss of dynamic range at certain frequencies or frequency bands. A variety of prior analog and digital hearing aids have been designed to mitigate the above-identified hearing deficiency with loss of dynamic range. To compensate for the loss of dynamic range, prior art hearing instruments have used a so-called multi-band dynamic range compressor to compress the dynamic range of the incoming sound such that the compressed output signal more closely matches the dynamic range of the intended user. The ratio of the input dynamic range to the dynamic range output by the multi-band dynamic range compressor is referred to as the compression ratio. Typically, the multi-band dynamic range compressor is configured to perform differently, e.g. different compression ratios and/or different attack and release time constants, in different frequency bands to accounting for the frequency dependent loss of dynamic range of the intended hearing impaired user.
U.S 2003/0081804 discloses a so-called side-branch architecture for a multi-band dynamic range compressor based on the Fast Fourier Transform (FFT). The multi-band dynamic range compressor uses a side branch for the frequency analysis of the audio input signal. The FFT is computed on a warped frequency scale from outlet taps of a cascade of first-order all-pass filters to which the audio input signal is applied. The same tapped delay line is used for both the FFT analysis and a time-varying FIR compression filter. Results of the FFT based frequency analysis are used to generate the coefficients of the FIR compression filter placed in the signal path.
The warped frequency scale and side-branch architecture of the disclosed multi-band dynamic range compressor result in numerous desirable properties such as minimal time delay as the direct signal path contains only a short input buffer and the FIR compression filter. Other noticeable advantages are absence of aliasing and a natural log-scaling of the analysis frequency bands conforming nicely to the Bark based frequency scale of human hearing. However, the disclosed FFT-based multi-band dynamic range compressor suffers from certain undesired properties. In particular, signal spectrum values of all frequency bands of the FFT based analysis are updated at the same block rate or frequency which may lead to undersampling of high frequency components of the input sound. Undersampling of the high frequency components is generally undesirable as it may cause aliasing of spectral level estimates in the analysis frequency bands and result in misbehaving and distortion inducing compression gain agents or coefficients.
Furthermore, while a relatively high block rate may be selected in the FFT based multi-band dynamic range compressor to accommodate the high frequency components, this will lead to a faster update of low frequency bands of the analysis filter than required for correct sampling, i.e. oversampling of the low frequency bands. While the latter oversampling property does not cause aliasing distortion, it wastes computational resources of a signal processor of the hearing instrument executing the FFT-based multi-band dynamic range compressor. This process incurs unnecessary power consumption by the hearing instrument which shortens the battery life time.